Jason D. Wolfe
2005-Dec-25 11:07 UTC
[Asterisk-Users] newbie question about making outbound call
Hello, Somehow I've missed something here, so hopefully I'll be able to provide enough of my setup to get some help. I feel I'm very close to getting it, but missing something none the less... 1. I have a digium TDM400 with (2) FXO modules on channel 3 and 4 hooked to two POTS lines. 2. I have the following entry in zapata.conf file: usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes callprogress=no context=incoming signalling=fxs_ks channel=>4 3. I have the following entry in extensions.conf [callAgent] exten=>outbound,1,Dial(Zap/4/phonenumber) ;where phonenumber is a 10 digit number exten=>outbound,n,Playback(access-code) ; just for the sake of doing something! 4. I am using Asterisk Java Manager AGI OriginateAction with the following code in a jsp page running on a tomcat server: //manageAGI ManagerConnection managerConnection; ManagerConnectionFactory factory; OriginateAction originateAction; ManagerResponse originateResponse; factory = new ManagerConnectionFactory(); managerConnection = factory.getManagerConnection("192.168.1.4","jason", "nosaj111"); // connect to Asterisk and log in managerConnection.login(); originateAction = new OriginateAction(); originateAction.setAsync(true); originateAction.setChannel("Zap/4"); originateAction.setContext("callAgent"); originateAction.setExten("outbound"); originateAction.setPriority(new Integer(1)); originateAction.setTimeout(3000); originateResponse = managerConnection.sendAction(originateAction, 30000); 6. when I execute the jsp page, I watch the console started with /usr/sbin/asterisk -cvvvvvvvvvv and I get the following message (I substituted phonenumber in for the 10 digit number again) *CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'jason' logged on from 192.168.1.3 > Channel Zap/4-1 was answered. -- Executing Dial("Zap/4-1", "Zap/4/phonenumber") in new stack Dec 25 10:55:40 NOTICE[3989]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Playback("Zap/4-1", "access-code") in new stack -- Playing 'access-code' (language 'en') == Manager 'jason' logged off from 192.168.1.3 == Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/4-1' exten => outbound,1,Hangup() What I eventually want to accomplish is the following: I want a web user (using a JSP page I think) to be able to click a button and cause asterisk to dial outbound on both FXO ports, wait for an answer, play some files, accept some input, and bridge the two calls together. am I on the wrong track? is there anything that is standing out that I am just not understanding here? ANY comments will be much appreciated. Thank you, Jason
Moises Silva
2005-Dec-27 08:36 UTC
[Asterisk-Users] newbie question about making outbound call
Hi Jason. It seems your doing things "right" whatever that means. I think the problem is more hardware related. Sure you have line in the FXO?? have you tried dialing directly from some IP Phone?? I have several applications that relay on automatic call generation with Asterisk Manager and a PHP classes i have. But, as i said, i think the problem is related to the configuration of the card. what does ztcfg -vv says? what does zttool says?? best regards On 12/25/05, Jason D. Wolfe <jason_d_wolfe@comcast.net> wrote:> > Hello, > > Somehow I've missed something here, so hopefully I'll be able to provide > enough of my setup to get some help. I feel I'm very close to getting > it, but missing something none the less... > > 1. I have a digium TDM400 with (2) FXO modules on channel 3 and 4 hooked > to two POTS lines. > 2. I have the following entry in zapata.conf file: > > usecallerid=yes > hidecallerid=no > callwaiting=no > threewaycalling=yes > transfer=yes > echocancel=yes > echotraining=yes > callprogress=no > context=incoming > signalling=fxs_ks > channel=>4 > > 3. I have the following entry in extensions.conf > > [callAgent] > exten=>outbound,1,Dial(Zap/4/phonenumber) ;where phonenumber is a 10 > digit number > exten=>outbound,n,Playback(access-code) ; just for the sake of doing > something! > > 4. I am using Asterisk Java Manager AGI OriginateAction with the > following code in a jsp page running on a tomcat server: > > //manageAGI > ManagerConnection managerConnection; > ManagerConnectionFactory factory; > OriginateAction originateAction; > ManagerResponse originateResponse; > > factory = new ManagerConnectionFactory(); > managerConnection = factory.getManagerConnection("192.168.1.4","jason", > "nosaj111"); > > // connect to Asterisk and log in > managerConnection.login(); > > originateAction = new OriginateAction(); > originateAction.setAsync(true); > originateAction.setChannel("Zap/4"); > originateAction.setContext("callAgent"); > originateAction.setExten("outbound"); > originateAction.setPriority(new Integer(1)); > originateAction.setTimeout(3000); > > originateResponse > managerConnection.sendAction(originateAction, 30000); > > > 6. when I execute the jsp page, I watch the console started with > /usr/sbin/asterisk -cvvvvvvvvvv > and I get the following message (I substituted phonenumber in for the 10 > digit number again) > > *CLI> == Parsing '/etc/asterisk/manager.conf': Found > == Manager 'jason' logged on from 192.168.1.3 > > Channel Zap/4-1 was answered. > -- Executing Dial("Zap/4-1", "Zap/4/phonenumber") in new stack > Dec 25 10:55:40 NOTICE[3989]: app_dial.c:1010 dial_exec_full: Unable to > create channel of type 'Zap' (cause 0 - Unknown) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing Playback("Zap/4-1", "access-code") in new stack > -- Playing 'access-code' (language 'en') > == Manager 'jason' logged off from 192.168.1.3 > == Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL' > -- Hungup 'Zap/4-1' > exten => outbound,1,Hangup() > > What I eventually want to accomplish is the following: > > I want a web user (using a JSP page I think) to be able to click a > button and cause asterisk to dial outbound on both FXO ports, wait for > an answer, play some files, accept some input, and bridge the two calls > together. > > am I on the wrong track? is there anything that is standing out that I > am just not understanding here? ANY comments will be much appreciated. > > Thank you, > Jason > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051227/64941982/attachment.htm